Traditional telephone service is circuit-switched—every call is transmitted over dedicated facilities reserved for that particular call. Long distance calls are transmitted from the user's phone line over copper wires to a local telephone company's network switch, which converts the call to digital format and hands it off to the long distance carrier. The long distance carrier then routes the call over its network to a local telephone company's network serving the call recipient. That carrier's local network switch re-converts the call to analog signals and connects the call to a dedicated line serving the recipient. As long as the call is ongoing, a circuit—a dedicated splice of bandwidth—remains open throughout all three networks involved in transmitting the voice signals. Both the called party and the calling party pay a fee to their local telephone company for access to each company's local network, and the calling party pays an additional fee, usually on a per minute basis, for use of the long distance carrier's network.
Internet-based telephone services, or IP telephony, presently offer significant benefits over traditional telephone service. Although users generally pay a fee for Internet access, it is usually a set amount on a monthly basis regardless of the amount of bandwidth used. If two people use the Internet to call one another, they can by-pass both local telephone companies' networks and the long distance carrier's network, without incurring additional fees for making such a call. This type of IP telephony, that does not require any intermediary service (charged at some price), is extremely inconvenient because it requires both users to be using the same software at the same time and to know that the other user is available on-line at the designated time of the call. Other technologies have been developed to enhance the convenience of calling, but services utilizing these technologies often require additional hardware devices to be installed on each user's system and have introduced other issues, such as service quality, which will be discussed more fully below.
Internet service providers are naturally interested in IP telephony because IP telephony increases the demand for access services. Customers have been reluctant to use IP telephony, however, because the quality of such services has not been as high as standard telephony and there have been significant limitations on who could be called using IP telephony. Nevertheless, Internet Telephony Service Providers (“ITSP”) are realizing there is a significant marketing advantage associated with having the ability to offer IP telephony. Since some IP telephony services are of a lower quality than traditional telephony, this provides ITSP's a way to version their service on quality: high quality/high price and low quality/low price, with variations in between being developed. By offering multiple versions of their product, ITSPs can better match their products against their customer's needs and their willingness to pay. This enables them to extract the full value of their services from their customers.
A Voice over IP (“VoIP”) system enables the transmission of telephone calls over an IP data network such as the Internet. A VoIP system handles a telephone call over most of the network as just another stream of data. Typically, an IP telephony user dials a toll-free number to connect the user to an IP telephony gateway. The gateway is the key element here, as it bridges the public telephone network and the public or private IP network providing the service. Once connected to the gateway, the user dials his or her account number (for billing purposes) and the destination phone number of the call. The gateway receives telephone signals on one side, converts them to IP packets, and outputs the packets to public or private IP networks for routing to the terminating user, and vice versa. A typical packet includes 10 to 30 milliseconds worth of conversation. Each packet is coded with the second party's phone number, and compressed for rapid transmission.
The packets travel the IP network, passing through routers, computers that operate like switches by reading the addresses on each packet and assigning them to appropriate transmission lines, to arrive at a gateway that decompresses them and converts the packets back into a voice transmission signal. The gateway then passes the call to the local phone network, which delivers it to the intended party.
One of the key challenges emerging from the integration between circuit switched (traditional phone) and packet switched (IP) networks is how to address calls that pass from one network service to another. Currently, it is possible to originate calls from IP address-based networks to other networks, however it is difficult to terminate calls from other networks, such as the PSTN, to IP address-based networks. Instead, calls from the PSTN are typically being terminated on the PSTN. For example, in PCT Application Number PCT/US99/29171, published as International Publication Number WO 00/41383, by Ranalli et al., a system is described for resolving a PSTN number to an IP address for voice communication between two simultaneous users over the Internet, but which system lacks an IP-enabled gateway for completing calls from an Internet user to a PSTN phone or from a PSTN phone to an Internet user. Hence, a calling party must use a standard terminal device, such as a phone, for connecting to another user on the PSTN. Therefore, there is a need for an addressing system across both PSTN and public packet networks that can allow a PSTN caller to access a subscriber on a public packet network and vice versa.
Although intermediate gateways exist, they present another problem in prior VoIP systems. As noted above, in some systems, a user is required to call a gateway, which then calls the recipient's number. This results in over dialing and may cause some calls to be blocked at the receiving end. Further, in other prior systems, the mapping data for logged on subscribers is not published in a public forum that can be accessed by other subscribers and gateways connected to PSTN terminals. Rather, the providers of such systems distribute a hardware device to subscribers only, who use that device to pull mapping data for other subscribers from a database server. Therefore there is a need for a system that handles the entire call without a central answering position (an over dialing gateway) and which allows any user, even if through an intermediary, easy public access to subscriber mapping data. The present invention allows subscribers to call other subscribers and completely bypass the PSTN local office by using a call agent browser to connect to a gateway device and to send mapping data and receive mapping data from other users via distributed servers. The call agent can be an applet or any specifically enabled Internet browser. The call agent can also support soft-phone software applications, provide a user profile, and other user information.